1. Field of the Invention
The present invention relates to a telephone system employing Internet Protocol (IP), and in particular to a sound output setting system for use in information processing terminals having a telephone terminal function and capable of easily setting and updating a received audio signal. The audio signal is a call progress tone signal, such as a signal of a dial tone (DT), ring-back tone (RBT), or busy tone (BT), received by the terminal.
2. Description of the Related Art
Currently, the conventional circuit-switching network architectures are being rapidly replaced with Internet architectures. This consequently increases the use of Session Initiation Protocol (SIP) as a communication protocol for Internet telephones employing Voice over IP (VoIP). The SIP is a communication protocol enabling a user to freely select a desired application or service by means of end-to-end control and mild and flexible management of networks and services. Some office telephone systems use a private branch exchange or key telephone system to form a communication network. However, there are some services among the existing services, that cannot be provided as a unified service in these systems by using the end-to-end control architecture.
In these existing services, call progress tones provided to users are included. The call progress tones are provided to a user prior to establishment of a media session at a telephone terminal. In the case of a circuit-switching network architecture, the services are managed by a centralized controller possessed by a telephone exchange for the purpose of controlling the same. This means that an audio channel is formed between the centralized controller and the telephone terminal prior to establishment of the session between the telephone terminals. Therefore, the user is allowed to hear a call progress tones such as a dial tones (DT) or ring-back tone (RBT) through a signal sound generated by the centralized controller. In the case of an end-to-end control architecture such as SIP, however, the call control is performed by means of a text type message through the communication protocol. Therefore, no audio channel is established until a called party responds to the call and a talk session is established. Consequently, the centralized controller does not intervene during the talk session. The call progress tone thus has to be generated by the telephone terminals. The telephone terminals are allowed to select freely a type of tones to be output. However, unified specifications are applied to design the exchanges and the telephone terminals. Thus, a complicated processing is required to unify the specifications thereof if the tones to be transmitted are to be unified in the system.
In order to solve such problems, office telephone system using SIP as a communication protocol employ an extended Back-to-Back User Agent (B2BUA) in which a SIP User Agent (SIPUA) is used as a central telephone equipment having an exchange function or a telephone exchange. According to this method, the telephone exchange communicates with a calling terminal and a called terminal while connecting their audio channels in the telephone exchange. In this case, a talk session is established between the telephone exchange and the telephone terminal, and a call progress tone formed by output sounds generated by the telephone exchange is heard at the telephone terminal. However, the SIP session state of the telephone terminal is different from the actual communication session state, which substantially reduces the design freedom of client applications using the SIP.
An improvement for solving this problem is disclosed, for example, in JP-2004-312357-A2 (Patent Document 1) relating to an Internet telephone system. According to the Patent Document 1, a telephone unit connected to the Internet receives streaming delivery of audio data from an audio server connected to the Internet, and a telephone terminal outputs the audio data. Using the streaming delivery, however, packet arrival delay is apt to occur when the network traffic is high. Further, troubles may occur such as clipping of the beginning of the tone sound and interruption of the tone sound when streaming delivery processing load is increased due to concentration of accesses to the audio server.
In order to solve such problem of output sound interruption, a system can be packaged to be composed of central IP telephone equipment serving as a telephone exchange and IP telephone terminals. This is effective means for manufacturers. Specifically, in the system, the telephone terminal has a digital signal processor (DSP) mounted thereon, and when a communication control command is issued by the IP telephone exchange to the IP telephone terminals, the communication control command notifies parameter information to form the output sound. This processing enables the IP telephone terminals to dynamically produce and output the related sound. A management table of audio files corresponding to the sound output information notified by the IP telephone exchange and all the prospective audio files must be prepared in the IP telephone terminals.
However, the ringing specifications such as frequencies and cycles of the call progress tones differ among countries and regions. Therefore, the output sound specifications have to be changed according to a country or region where the IP telephone exchange is installed. The IP telephone terminal has to restructure the audio files into an application compliant to the standard of the country or region according to the output sound specifications of the IP telephone exchange. More specifically, the IP telephone application provides an install image containing an audio file to the terminal user. Further, in some cases, the providing vendor of the IP telephone exchange is different from that of the IP telephone application. If this is the case, whenever a vendor providing the IP telephone exchange changes the specifications, a vendor providing the IP telephone application is following the change and requires to prepare a restructured install image.
Further, the IP telephone terminal has no audio files corresponding to the output sounds. Therefore, if the IP telephone exchange adds new sound output information and the IP telephone terminal receives a communication control command to output the new output sound, the IP telephone terminal is not able to output the related sound. In order to avoid such circumstances, it is known to provide preliminary sound output information so that the new audio file is provided to the IP telephone application vendor to restructure the install image. However, this method is restricted if the prepared preliminary sound output information is limited. It is in the first place impossible to solve the problem of occurrence of situation in which the IP telephone application vendor is required to restructure the install image whenever a new audio file is added.
The present invention relates to a system in which central telephone equipment having a telephone exchange function using IP is combined with telephone terminals via a network, and communication is controlled by Voice over IP (VoIP). In such a system, output sound specifications for call progress tones need be changed according to a country or region where the central telephone equipment is installed. However, the change of the output sound specifications cannot be performed easily by simple configuration. The present invention intends to solve such problems of the system.